On 26 Mar 2014, at 16:32, Mickael MONSIEUR &lt;mickael.monsieur(a)gmail.com&gt; wrote: Hello, I installed kamailio and asterisk with the tutorial of asipto. For alias numbers I configured the module alias_db. Everything works because Asterisk outgoing call is directed at Kamailio, then Kamailio is sending to the alias registered (table location) Only the number alias appears in the "To" (02XXXXXX) and not in the INVITE URI. It only shows "s" ... is it possible to force writing 02XXXXXX instead of "s" ? Who sets the "s" in the request URI? Asterisk use "s" when there's no extension given in the dialplan. /O Example: INVITE sip:s@10.1.0.191:5060 SIP/2.0 Max-Forwards: 69 From: "0475XXXXXX" <sip:1053212@sip.domain.com>;tag=as7df9ab18 To: <sip:02XXXXXX@kamailioIP:5060> Contact: <sip:1053212@asteriskIP:5060> Call-ID: 344d42bd16975a54141d11f635bdfc71(a)sip.domain.com CSeq: 102 INVITE Date: Wed, 26 Mar 2014 15:06:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 252 Because without it, Asterisk servers behind Kamailio not will route the call to the correct extension but to the "s". Asterisk ignores the "To" apparently this is strange ... Thank you, Mickael _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users